polycom sip

Release date: 2012-04-23Updated on: 2012-04-24 Affected Systems:Asterisk 10.xAsterisk 1.xUnaffected system:Asterisk 10.3.1Asterisk 1.8.11.1Description:--------------------------------------------------------------------------------Bugtraq id: 53205

SipManager: setautoregisterandpolicy () --> SipService: openpolicysession () --> SipSessionGroupExt: opentoreceivecballs () opentoreceivenovel () --> AutoRegistrationProcess: start () --> first, perform the anti-registration (duration = 0)

If You know this phone is ringing (an ALERT q.931 message, for instance) you send a ringing. If You receive a notification indicating then the call was progressing, but don't know for sure whether the user I s being alerted or not, you send a 183

Release date: 2011-10-18Updated on: 2011-10-18 Affected Systems:Asterisk Open Source 10.xAsterisk Open Source 1.8.xDescription:--------------------------------------------------------------------------------Cve id: CVE-2011-4063 Asterisk is a free

The RTP receiving part is relatively simple (you do not need to consider jitterbuffer and so on). Start with here. In fact, there are three steps: 1. Create a UDP listener, such as 5200. 2. After receiving the RTP package, send it to the unpacking

set portopening udp/tcp/all port_number name enable So, in order to save time and energy, create a new Excel document, in which to add statements, the following figure: Here, the name I give to the port is called the polycom-port number, then select all the records and copy them to the text document. and remove the tab character in the middle of the name. This step is easy, just to save some energy. If it's not too much trouble, you can write it all

follows:Netsh firewall set portopening udp/tcp/all port_number name enableTherefore, in order to save time and energy, first create an excel file and write the statements to be added, such: Here, I name the port polycom-port number, select all records, and copy them to the text document. and delete the TAB character in the middle of the name. this step is easy, just to save some effort. if it is not too troublesome, you can write it all into the text

Entry Point of NGN access control security -- Diameter protocol and its application in the SIP network environment Xie Wei I. Introduction The diameter series protocol is a new generation of AAA technology, which is gaining more and more attention due to its powerful scalability and security assurance. In international standards organizations such as ITU, 3GPP and PP2, DIAM-ETER protocols have been officially used as the preferred AAA protocol for fut

Document directory RTP Stacks (mainly open source C/C ++ stacks) SIP Stacks RTP Applications SIP Phones (SIP User Agents) SIP Test Utility SIP Applications (Proxy, Location Server) Sip Express Router (ser) Ser Media Serv

VoIP bookmarks from Klaus darilion Below you will find descriptions and links to sip and RTP stacks, applications, test utilities, SIP proxies, SIP pbxs and stun server and clients. most of them are open source :-), but not all of them If you have any comments please feel free to contact me: --> Klaus. darilion at pernau. at There are also other VoIP related por

Document directory Jain proposal SIP, ISUP, call control system, and Jain Interface Application of Jain APIs to Mobile Networks Mobile Station No-wire access to network (RAN) Network and Enterprise Internal capacity and service End-to-end structure Jain API for call control and Wireless Networks Face-to-Face Jain API of the integrated network connects the business agility, network convergence, and security network to the telephone and

calls can be interconnected, at the same time.Scenario Networking　　Programme descriptionCargosmart interfaces with Polycom video conferencing MCUs via the either an over-the-pass or SIP protocol. CyberLink uses the PRI to connect to the operator's telephone network. Alternatively, use the PRI, SIP, and Ethernet to connect to the enterprise's existing switches.　　

Comparison between H.323 and SIP Currently, 3GPP uses SIP as the core protocol of the third-generation mobile communication network. The NetMeeting protocol in Windows XP is also changed from H.323 to the SIP protocol. Considering its business flexibility, the SIP protocol will become the future development direction.

In terms of routes and switches, we have learned a lot about them. Now, let's take a look at the content about the softswitch protocol. Currently, 3GPP uses SIP as the core protocol of the third-generation mobile communication network. In Windows XP, NetMeeting's Softswitch protocol is also changed from H.323 to the SIP protocol. Considering its business flexibility, the

Microsoft, everyone on Earth knows that it is a representative of windows, so its strength is the operating system and software. Cisco is also not unknown. It is the manufacturer of network products. It is only unexpected hardware, and it does not produce hardware ...... The two companies have launched their own unified communication products. Now let's look at the differences between them. 1. Voice/online status display Microsoft's voice/online status display corresponds to Microsoft's OfficeCo

Reference: http://mbstudio.spaces.live.com/blog/cns! C898c3c401_dc11! 955. Entry For the latest version of this document and the relevant source code and vc6 engineering files mentioned in this article, please find them on this site ~~(In the skydriver public folder on the homepage, you may need to useProxyCan access the space normally-the space is absolutely stable and files will not be lost !) (The focus of recent work is not on SIP development, so

Session Initiation Protocol (SIP) is a control (signaling) Protocol for the application layer of Network calls and conferences. It is mainly a multimedia communication protocol based on an IP network. All the signaling functions it can implement also use RTP as the media transmission protocol. It was initially proposed by the IETF mmusic (multiparty multimedia session control) Working Group. The main functions of

When talking about the Unified Communication field, we are sure we know the contest between Cisco and Microsoft. From their own development perspective, Cisco = hardware, Microsoft = software. So two completely different methods bring different solution concepts to enterprises. What are their characteristics? Cisco and Microsoft have adopted completely different methods in the field of Unified Communication: Cisco's approach is based on network and hardware, while Microsoft's approach is "softwa

Author: gnuhpcSource: http://www.cnblogs.com/gnuhpc/ 1. Definition of VoIP: voice services with certain service quality transmitted over an IP network. 2. Key VoIP technologies: Speech Processing Technology Minimize the bit rate (voice encoding technology, voice detection and suppression technology) while ensuring a certain speech quality) Ensure certain call quality in the IP environment (packet loss compensation, Echo offset, and dejitter) Voice Communication Protocol Call Control Protoc

IP addresses.Other:DNS (domain Name Service), which is used to complete address lookups, mail forwarding, and so on (running on TCP and UDP protocols).Echo (Echo Protocol, wrap-around protocol) for error detection and measurement response time (running on TCP and UDP protocols).SNMP (Simple network Management Protocol), which is used for network information collection and network management.ARP, Address Resolution Protocol, addresses resolution protocol for dynamic resolution of Ethernet hardwa